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VoIP troubleshooting-Internet

VoIP is dependant on both the local and wide area
network. If voice quality problems occur, either for brief
periods or longer durations the problem can often be the direct result of the
network connection. This would include your ISP Internet Service Provider
or some other component between you and the party that you are talking with.

Some issues are probably not a network related problem. One would be
constant echo on your VoIP line, although a
network issue could make this echo worse over certain periods. Another
would be the inability not to connect to specific phone numbers. For other
consistent voice quality issues we recommend looking at out
VoIP tutorials first.

VoIP symptoms of network issues.

Many quality issues with your VoIP connection could
be the direct result of network problems. Although these issues
could be the result of something else other than network related problems, they
are highly indicative of one, especially if your VoIP connection works well most
of the time and then experiences problem periods. These symptoms are:

Delays between the time that you speak and when the other side hears you
words, (and vice versa).

Talking over each other because neither end realizes the other already
started speaking.

Gaps of missing voice.

Causes of poor quality VoIP.

The network related causes of poor quality VoIP as described by the symptoms
above, can be broken down into three main categories.

Packet Loss

Data packets carrying parts of a conversation can be lost during
transmission. If enough packets are lost, and VoIP is very sensitive to
and has a very low threshold to packet loss, then gaps or audible problems will
quickly be discernible. The protocols or codecs that are used to transport
voice often incorporate Packet Loss Concealment, which will help mask the
effects of lost or discarded packets, but they cannot overcome even low
percentages of packet loss.

Jitter

The difference in the transit time of packets varies and this difference is
called Jitter. As the packets of voice need to arrive in the same order as they
were sent, variation will result in a out of sequence part of sound which if not
placed in the proper order would make nonsense sound rather than distinguishable
words. An IP phone or ATA has a built in Jitter Buffer that introduces a small
amount of delay in order to smooth out and sequence these timing variations, but
only has a limited capability. (Actually this jitter buffer translates jitter
into additional delay and packet loss. Which is another reason VoIP is very
sensitive to Jitter.) If some packets arrive too late then they may be entirely
discarded producing the bitts of lost voice.

Latency

Latency or delay is the amount of time it takes data to travel from one
endpoint to the other endpoint. This travel time depends on several
factors including distance, queuing delay, which is the amount of time packets
are held in a queue because of congestion on an outbound interface of a device
(router), and handling delay which is the result of devices that forward the
frame through the network. VoIP depends on RTP traversing in a reasonable
amount of time (milliseconds), some people say typically no more than 150ms, but
conversations can still be good at the 200-220ms traverse time if other network
issues are not a problem (packet loss, jitter). VoIP users
will probably notice round-trip delays that exceed 250ms. Above 250ms and callers
will begin talking over each other. Around the 500ms range phone calls become
impractical and anyone carrying on a conversation with this amount of latency
had better end each sentence with "over".

How to network troubleshoot VoIP

To troubleshoot VoIP and the Internet or WAN network that the data is
traversing use a good free tool;
PingPlotter.
Using PingPlotter place the IP address of the providers SIP server as the target
address. The resulting graph can be used to locate packet loss at the end
targets IP.

Always look at the final destination first, and see if it is registering
packet loss. Then look at each upstream hop for the first one (closest to
the last hop) that does not show a similar loss. The packet loss problem
would then reside between the good hop and the bad hop (hand off).

If you're not seeing any packet loss at the final destination then the other
intermediate hops shouldn't be of any concern regarding packet loss. But
they could be introducing latency or delay at which point they would be of
concern.

Congestion
can be recognized by looking at a PingPlotter graph. This event can have
detrimental effects on VoIP as the congestion introduces latency and packet
loss. Some nodes on cable systems can be over subscribed and during peak
hours experince congestion. Typically peak hours would be between 6PM and
9PM.

If packet loss is accompanied by a high level of jitter then congestion could
be the problem. Or if packet loss comes and goes in large amounts (known
as bursty) then the problem might also be network congestion.